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	Secret Rabbit Code (aka libsamplerate)
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Author :<BR>Erik de Castro Lopo
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<H1><B>Frequently Asked Questions</B></H1>
<P>
<A HREF="#Q001">Q1 : Is it normal for the output of libsamplerate to be louder 
	than its input?</A><BR><BR>
<A HREF="#Q002">Q2 : On Unix/Linux/MacOSX, what is the best way of detecting
	the presence and location of libsamplerate and its header file using
	autoconf?</A><BR><BR>
<A HREF="#Q003">Q3 : If I upsample and downsample to the original rate, for
	example 44.1->96->44.1, do I get an identical signal as the one before the
	up/down resampling?</A><BR><BR>
</P>
<HR>
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<A NAME="Q001"></A>
<H2><BR><B>Q1 : Is it normal for the output of libsamplerate to be louder 
	than its input?</B></H2>
<P>
The output of libsamplerate will be roughly the same volume as the input. 
However, even if the input is strictly in the range (-1.0, 1.0), it is still 
possible for the output to contain peak values outside this range.
</P>
<P>
Consider four consecutive samples of [0.5 0.999 0.999 0.5]. 
If we are up sampling by a factor of two we need to insert samples between 
each of the existing samples. 
Its pretty obvious then, that the sample between the two 0.999 values should 
and will be bigger than 0.999.
</P>
<P>
This means that anyone using libsamplerate should normalise its output before
doing things like saving the audio to a 16 bit WAV file.
</P>

<!-- ========================================================================= -->

<a NAME="Q002"></a>
<h2><br><b>Q2 : On Unix/Linux/MacOSX, what is the best way of detecting
	the presence and location of libsamplerate and its header file using
	autoconf?</b></h2>

<p>
libsamplrate uses the pkg-config (man pkg-config) method of registering itself
with the host system.
The best way of detecting its presence is using something like this in configure.ac
(or configure.in):
</p>

<pre>
    PKG_CHECK_MODULES(SAMPLERATE, samplerate >= 0.0.15, 
            ac_cv_samplerate=1, ac_cv_samplerate=0)

    AC_DEFINE_UNQUOTED([HAVE_SAMPLERATE],${ac_cv_samplerate},
            [Set to 1 if you have libsamplerate.])

    AC_SUBST(SAMPLERATE_CFLAGS)
    AC_SUBST(SAMPLERATE_LIBS)
</pre>
<p>
This will automatically set the <b>SAMPLERATE_CFLAGS</b> and <b>SAMPLERATE_LIBS</b>
variables which can be used in Makefile.am or Makefile.in like this:
</p>
<pre>
        SAMPLERATE_CFLAGS = @SAMPLERATE_CFLAGS@
        SAMPLERATE_LIBS = @SAMPLERATE_LIBS@
</pre>

<p>
If you install libsamplerate from source, you will probably need to set the
<b>PKG_CONFIG_PATH</b> environment variable's suggested at the end of the
libsamplerate configure process. For instance on my system I get this:
</p>
<pre>
    -=-=-=-=-=-=-=-=-=-= Configuration Complete =-=-=-=-=-=-=-=-=-=-=-
    
      Configuration summary :
    
        Version : ..................... 0.1.0
        Enable debugging : ............ no
    
      Tools :
    
        Compiler is GCC : ............. yes
        GCC major version : ........... 3
    
      Extra tools required for testing and examples :
    
        Have FFTW : ................... yes
        Have libsndfile : ............. yes
        Have libefence : .............. no
    
      Installation directories :
    
        Library directory : ........... /usr/local/lib
        Program directory : ........... /usr/local/bin
        Pkgconfig directory : ......... /usr/local/lib/pkgconfig
</pre>


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<A NAME="Q003"></A>
<H2><BR><B>Q3 : If I upsample and downsample to the original rate, for
	example 44.1->96->44.1, do I get an identical signal as the one before the
	up/down resampling?</B></H2>
<P>
The short answer is that for the general case, no, you don't.
The long answer is that for some signals, with some converters, you will
get very, very close.
</P>
<P>
In order to resample correctly (ie using the <B>SRC_SINC_*</B> converters),
filtering needs to be applied, regardless of whether its upsampling or 
downsampling.
This filter needs to attenuate all frequencis above 0.5 times the minimum of
the source and desination sample rate (call this fshmin).
Since the filter needed to achieve full attemuation at this point, it has to
start rolling off a some frequency below this point.
It is this rolloff of the very highest frequencies which causes some of the
loss.
</P>
<P>
The other factor is that the filter itself can introduce transient artifacts
which causes the output to be different to the input.
</P>

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